August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

DTMF bleed through in conference
by David Bolen 17 Aug '11

17 Aug '11

I'm encountering what I would probably best describe as some bleed through of inbound DTMF when in a conference, and I was wondering if anyone might have suggestions for likely causes or troubleshooting? DTMF detection itself seems fine, and appears to properly be using rfc2833 (the SDPs all show include "a=rtpmap:101 telephone-event/8000"). I get "RTP RECV DTMF" logs for each digit that is recognized. However, a reasonable percentage of the time, some small fraction of the DTMF appears to be heard by the conference as a whole. Never the whole thing, and sometimes it's a bit distorted (though that may just be due to it being so short). Almost as if a small portion of a sound sample makes it through. For testing, I'm using internal phones with direct SIP connections to the switch so no PSTN or gateways involved, though it also occurs with a PSTN caller coming in through multiple gateways. For the internal phones, I've tried both a physical phone through a PAP2T, as well as a PC soft phone. This is with git-868d823 from Jul 29, so a few weeks old but reviewing the more recent commits there weren't any seemingly related to DTMF or conferencing. Any suggestions on places to look or things to try would be most appreciated. -- David

5 7

FS performance using ESL
by Tihomir Culjaga 16 Aug '11

16 Aug '11

hello, im wondering how much performance do we loose when using ESL instead of running it via dialplan? without ESL with a fine tuned FS and a short dialplan ( answer, playback like 20 seconds file, hangup ) im able to service 75 CPS. On the same FS, when i use ESL to answer the call, playback the same file and hangup, im not able to run more than 2 CPS... this is a huge impact and i really can't believe it. I'm using event-socket outbound e.g.: <action application="socket" data="127.0.0.1:8088 async full"/> my extension looks like: <extension name="ESL_C"> <condition field="destination_number" expression="^(6666)$"> <action application="socket" data="127.0.0.1:8088 async full"/> <action application="sleep" data="1000"/> <action application="hangup"/> </condition> </extension> im using testserver from lib/esl/ and i just removed the conference command and added the playback one.... also i moved the esl_debug lvl to 0 anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the playback is done from the dialplan. Please, can someone give me a clue on what is going on? Maybe im doing something wrong? how to get maximum FS performance using ESL ? Regards, Tihomir.

2 8

Building Freeswitch + FreeTDM + LibSng-ISDN RPMs
by Octavio Ruiz 16 Aug '11

16 Aug '11

I'm trying to build Freeswitch with full Sangoma's ( FreeTDM + LibSng-ISDN ) support from SRPMs for Centos 5, but there are two not satisfied dependencies for what looks like non existing RPMs, fact that raises many questions on my head, An extract from the freeswitch SPEC file: -- from freeswitch.spec Requires: libsng_isdn BuildRequires: wanpipe BuildRequires: libsng_isdn even if looks quite easy to bypass this and get it to compile satisfying those, either creating the missing libsng_isdn SPEC file needed for building the package (libsng_isdn is only a bunch of headers and an already-build shared object) and modifying the wanpipe.rpmspec in order to change the naming behavior which doesn't comply with the above (Wanpipe's resulting RPM is named 'wanpipe-<kernel-version>' which do not satisfy the solely string 'wanpipe') or either worst, in terms of cleanliness, creating dummy RPM packages or removing dependencies on the freeswitch.spec, I wonder what is supposed to be the correct environment and proper procedure for building Freeswitch and Wanrouter + SNG-ISDN following what the SPEC file states. -- from wanpipe.rpmspec %define KERNEL_VERSION %{?kern_ver} %define WANPIPE_VER wanpipe %define name %{WANPIPE_VER} Am I missing something? Has anyone tested this before? Someone else is packaging Sangoma's Wanrouter on RPM with other naming scheme? Are they on the SPEC file merely to denote upcoming work?

2 1

ORIGINATE_DISPOSITION
by Sam 16 Aug '11

16 Aug '11

For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?

5 20

any IVR example in C/C++?
by xl127 16 Aug '11

16 Aug '11

Hi, I've seen some examples for ASR/TTS in JavaScripts/Lua/Python etc. I can set my extension in the dialplan and point to my application e.g. by <action application="python" data="mySpeechApp.py"/>. I am wondering how I could do this for a C/C++ application? And in the scripts languages I can set a callback method, e.g. session.setInputCallback(myInputCallback) but I didn't find how to do this in C/C++. Googled around, but didn't find much helpful info on it. Any suggestions please? Thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278.

6 10

Lua not playing wav files
by Yuriy Nasida 16 Aug '11

16 Aug '11

Hi Freeswitch-users, My simple lua script: freeswitch.consoleLog("err","start hello.lua\n") session:answer(); message = "ivr/ivr-enter_destination_telephone_number.wav" session:execute("playback", message) session:hangup(); Script looks fine I think, but FS doesn't play audio. If I use corresponding XML dialplan all work fine. logs when I use lua: 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785@65.98.107.130:5080) State EXECUTE going to sleep 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785@65.98.107.130:5080) State ROUTING 2011-08-16 13:18:00.336003 [DEBUG] mod_sofia.c:147 sofia/external/79213777785@65.98.107.130:5080 SOFIA ROUTING 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785@65.98.107.130:5080 Standard ROUTING 2011-08-16 13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public 2011-08-16 13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785@65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits 2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101 2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785@65.98.107.130:5080] 65.98.107.130 port 26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20 2011-08-16 13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms 2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101 2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101 2011-08-16 13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785@65.98.107.130:5080: v=0 o=FreeSWITCH 1313488814 1313488815 IN IP4 65.98.107.130 s=FreeSWITCH c=IN IP4 65.98.107.130 t=0 0 m=audio 26266 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2011-08-16 13:18:00.348214 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785@65.98.107.130:5080 entering state [completed][200] 2011-08-16 13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application playback Requires media! pre_answering channel sofia/external/79213777785@65.98.107.130:5080 2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup 2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:988 sofia/external/79213777785@65.98.107.130:5080 destroy/unlink session from object 2011-08-16 13:18:00.348214 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785@65.98.107.130:5080) State ROUTING going to sleep" ======================================== logs when I use XML dialplan: 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785@65.98.107.130:5080) State EXECUTE going to sleep 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785@65.98.107.130:5080) State ROUTING 2011-08-16 13:19:49.609765 [DEBUG] mod_sofia.c:147 sofia/external/79213777785@65.98.107.130:5080 SOFIA ROUTING 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785@65.98.107.130:5080 Standard ROUTING 2011-08-16 13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785@65.98.107.130:5080) State ROUTING going to sleep 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785@65.98.107.130:5080) State EXECUTE 2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785@65.98.107.130:5080 SOFIA EXECUTE 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785@65.98.107.130:5080 Standard EXECUTE 2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785@65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785@65.98.107.130:5080] 65.98.107.130 port 23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20 2011-08-16 13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms 2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101 2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101 2011-08-16 13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785@65.98.107.130:5080: v=0 o=FreeSWITCH 1313491243 1313491244 IN IP4 65.98.107.130 s=FreeSWITCH c=IN IP4 65.98.107.130 t=0 0 m=audio 23946 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2011-08-16 13:19:49.619008 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785@65.98.107.130:5080 entering state [completed][200] 2011-08-16 13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16@8000hz 1 channels 20ms 2011-08-16 13:19:49.740067 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785@65.98.107.130:5080 entering state [ready][200] 2011-08-16 13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file 2011-08-16 13:19:52.880333 [NOTICE] switch_core_state_machine.c:189 sofia/external/79213777785@65.98.107.130:5080 has executed the last dialplan instruction, hanging up. I have compared logs and saw that case without lua have some strings unlike case with lua. "2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785@65.98.107.130:5080) State ROUTING going to sleep 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785@65.98.107.130:5080) State EXECUTE 2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785@65.98.107.130:5080 SOFIA EXECUTE 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785@65.98.107.130:5080 Standard EXECUTE" Many thanks to anyone who can help.

1 1

Billing solutions
by Cesar Bermudez 16 Aug '11

16 Aug '11

Hi FS users. I want to know, what billing solutions are available for FS. Open Source and not open source. Features, etc. Best regards

2 1

Notify message to Gateway
by Marc de Corny 16 Aug '11

16 Aug '11

Hi all, I have seen a lot of information on how to send NOTIFY messages to registered endpoints for ip phone resync. I'm trying to send the same notify messages over a sip trunk to another platform where the endpoints are registered. By putting the ip address of the remote server on the sip trunk in the host field of the send event FS did not send any messages. Am i missing anything? Do have to set the profile to external? Any infornation is helpful. Many thanks Marc

1 0

[Newbie] Unable to play .mp3 file
by Dhairya Vora 16 Aug '11

16 Aug '11

I have just tried a simple mp3 file, but freeswitch gives an error when call is established... freeswitch@internal> 2011-08-16 08:58:51.464482 [NOTICE] sofia.c:5725 Channel [sofia/external/00919879459879] has been answered 2011-08-16 08:58:51.464482 [NOTICE] switch_ivr.c:1663 Transfer sofia/external/00919879459879 to inline[socket:127.0.0.1:8084 async full@default] 2011-08-16 08:58:51.464482 [INFO] switch_channel.c:2638 sofia/external/00919879459879 Flipping CID from "" <00919879459879> to "Outbound Call" <gateway/flowroute/00919879459879> 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:806 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627. 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:809 Error from mpg123: Invalid mpg123 handle. (code 10) 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:806 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627. 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:809 Error from mpg123: Invalid mpg123 handle. (code 10) 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:806 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627. 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:809 Error from mpg123: Invalid mpg123 handle. (code 10) 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:806 Error: MPG123 Error at /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627. 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:809 Error from mpg123: Invalid mpg123 handle. (code 10) 2011-08-16 08:58:52.844500 [NOTICE] mod_dptools.c:916 Hangup sofia/external/00919879459879 [CS_RESET] [NORMAL_CLEARING] 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1347 Session 3 (sofia/external/00919879459879) Ended 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1349 Close Channel sofia/external/00919879459879 [CS_DESTROY] I checked that mod_shout is enable. The .mp3 file is not currupted. I am able to play in mp3 player. Please help solving this error: -Dhairya P.S. I am using plivo php file to make this call

2 3

getting disconnect cause for a leg after bridge in Lua
by Mi Ke 16 Aug '11

16 Aug '11

Hi All, Is there any way to get a real disconnection cause for leg B in the following script ? if (session_a:ready() and session_b:ready()) then freeswitch.bridge(session_a,session_b) -- session_b gets disconnect here ... local session_b_hangup_cause = session_b:hangupCause() session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ? Thanks / Mike

2 1